Services that perform peer-to-peer communication on the Internet require a technique for transmitting, in real-time, media information such as audio, video, text, and the like using IP (Internet protocol) packets. In addition, a signaling technique for controlling establishment, modification, and disconnection (termination) of a peer-to-peer session (call) on the Internet is essential. Peer-to-peer communication services include IP telephony, videotelephony, instant messaging, and the like.
In particular, IP telephony, whose popularity has increased dramatically in recent years, is realized as a combination of a VoIP (Voice Over IP) technique that enables real-time transmission of audio signals in IP packets, and a signaling technique.
As a signaling protocol usable in IP telephony, the H.323 standard recommended by the ITU-T (International Telecommunication Union-Telecommunication sector) in 1997 has been realized. In addition, SIP (Session Initiation Protocol) that was specified as a standards track in RFC 3261 standardized by the IETF (Internet Engineering Task Force) and issued in 2002 has been realized. In particular, in SIP, messages are described in text. Furthermore, SIP is designed based on HTTP (Hyper Text Transfer Protocol) for the Web and on SMTP (Simple Message Transfer Protocol) for e-mails. Consequently, SIP is simple, highly scalable, and has high affinity with the Internet. As a result, SIP is becoming the standard of signaling protocols used in IP telephony.
SIP is a signaling protocol for controlling establishment, modification, and disconnection (termination) of a session (call) between terminals on an application layer.
A method (SIP request message) and a response (SIP response message) are exchanged according to a predetermined procedure between terminals via a relay server called an SIP server which is disposed on the Internet. Accordingly, the establishment, modification, and disconnection (termination) of a session are controlled.
For example, establishing a session begins with a transmission of an INVITE message. In SIP, a terminal (UA: User Agent) is identified using a URI (Uniform Resource Identifier) format such as sip:hanako@fujitsu.com and sip:taro@fujitsu.com.
The exchange of an SIP message involves notifying what is used as a medium (audio, video, text). For example, in the case of an audio medium, an encoding scheme used, a protocol for carrying audio packets, a port number to be used, an audio packet transmitting frequency, and the like are notified.
In telephone services overall, responding to occurrences of network failure and promptly restoring service thereafter are essential for securing customer satisfaction and service credibility. When a failure occurs, swiftly comprehending network quality and an affected range (affected users) is required in order to appropriately respond to envisioned customer inquiries, to prevent the effect from spreading, and to restore service.
A conventional public switched telephone network (PSTN) uses a scheme in which communication paths are secured on a single line-basis, and a detection of a communication error, a congestion, or the like leads to the blockage of an entire communication path. Therefore, line quality and an affected range can be identified by verifying the blockage state of the communication path.
IP telephony is a new form of telephone service and is fundamentally a service in accordance with the IP protocol. Therefore, a technique that identifies network quality and a failure range based on the concept of a communication path, as is the case in PSTN, cannot be applied.
Conventionally, quality monitoring and failure detection in an IP telephone service depends on the quality of the IP network thereof and on a failure detecting method used in the IP network.
For example, Japanese Patent Laid-Open No. 2005-102180 discloses that a path combination in which a failure has occurred can be identified by measuring communication quality information on all path combinations that make up the Internet between terminals.